Packet-switched networks route data from a source to a destination in packets. A packet is a relatively small sequence of digital symbols (e.g., several tens of binary octets up to several thousands of binary octets) that contains a payload and one or more headers. The payload is the information that the source wishes to send to the destination. The headers contain information about the nature of the payload and its delivery. For instance, headers can contain a source address, a destination address, data length and data format information, data sequencing or timing information, flow control information, and error correction information.
A packet's payload can consist of just about anything that can be conveyed as digital information. Some examples are e-mail, computer text, graphic, and program files, web browser commands and pages, and communication control and signaling packets. Other examples are streaming audio and video packets, including real-time bi-directional audio and/or video conferencing. In Internet Protocol (IP) networks, a two-way (or multipoint) audio conference that uses packet delivery of audio is usually referred to as Voice over IP, or VoIP.
VoIP packets are transmitted continuously (e.g., one packet every 10 to 60 milliseconds) between a sending conference endpoint and a receiving conference endpoint when someone at the sending conference endpoint is talking. This can create a substantial demand for bandwidth, depending on the codec (compressor/decompressor) selected for the packet voice data. In some instances, the sustained bandwidth required by a given codec may approach or exceed the data link bandwidth at one of the endpoints, making that codec unusable for that conference. And in almost all cases, because bandwidth must be shared with other network users, codecs that provide good compression (and therefore smaller packets) are widely sought after.
Usually at odds with the desire for better compression is the desire for good audio quality. For instance, perceived audio quality increases when the audio is sampled, e.g., at 16 kHz vs. the eight kHz typical of traditional telephone lines. Also, quality can increase when the audio is captured, transmitted, and presented in stereo, thus providing directional cues to the listener. Unfortunately, either of these audio quality improvements roughly doubles the required bandwidth for a voice conference.